1/*
2 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/modules/audio_device/fine_audio_buffer.h"
12
13#include <limits.h>
14#include <memory>
15
16#include "testing/gmock/include/gmock/gmock.h"
17#include "testing/gtest/include/gtest/gtest.h"
18#include "webrtc/base/scoped_ptr.h"
19#include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
20
21using ::testing::_;
22using ::testing::AtLeast;
23using ::testing::InSequence;
24using ::testing::Return;
25
26namespace webrtc {
27
28// The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
29// to detect errors. This function verifies that the buffers contain such data.
30// E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
31// buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
32// will happen.
33// |buffer| is the audio buffer to verify.
34bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
35  int start_value = (buffer_number * size) % SCHAR_MAX;
36  for (int i = 0; i < size; ++i) {
37    if (buffer[i] != (i + start_value) % SCHAR_MAX) {
38      return false;
39    }
40  }
41  return true;
42}
43
44// This function replaces the real AudioDeviceBuffer::GetPlayoutData when it's
45// called (which is done implicitly when calling GetBufferData). It writes the
46// sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
47// buffer of different size than the one VerifyBuffer verifies.
48// |iteration| is the number of calls made to UpdateBuffer prior to this call.
49// |samples_per_10_ms| is the number of samples that should be written to the
50// buffer (|arg0|).
51ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
52  int8_t* buffer = static_cast<int8_t*>(arg0);
53  int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
54  int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
55  for (int i = 0; i < bytes_per_10_ms; ++i) {
56    buffer[i] = (i + start_value) % SCHAR_MAX;
57  }
58  return samples_per_10_ms;
59}
60
61// Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
62// for details.
63void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
64  int start_value = (iteration * size) % SCHAR_MAX;
65  for (int i = 0; i < size; ++i) {
66    buffer[i] = (i + start_value) % SCHAR_MAX;
67  }
68}
69
70// Action macro which verifies that the recorded 10ms chunk of audio data
71// (in |arg0|) contains the correct reference values even if they have been
72// supplied using a buffer size that is smaller or larger than 10ms.
73// See VerifyBuffer() for details.
74ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
75  const int8_t* buffer = static_cast<const int8_t*>(arg0);
76  int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
77  int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
78  for (int i = 0; i < bytes_per_10_ms; ++i) {
79    EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
80  }
81  return 0;
82}
83
84void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
85  const int kSamplesPer10Ms = sample_rate * 10 / 1000;
86  const int kFrameSizeBytes =
87      frame_size_in_samples * static_cast<int>(sizeof(int16_t));
88  const int kNumberOfFrames = 5;
89  // Ceiling of integer division: 1 + ((x - 1) / y)
90  const int kNumberOfUpdateBufferCalls =
91      1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
92
93  MockAudioDeviceBuffer audio_device_buffer;
94  EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
95      .WillRepeatedly(Return(kSamplesPer10Ms));
96  {
97    InSequence s;
98    for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
99      EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
100          .WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
101          .RetiresOnSaturation();
102    }
103  }
104  {
105    InSequence s;
106    for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
107      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
108          .WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
109          .RetiresOnSaturation();
110    }
111  }
112  EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
113      .Times(kNumberOfUpdateBufferCalls - 1);
114  EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
115      .Times(kNumberOfUpdateBufferCalls - 1)
116      .WillRepeatedly(Return(kSamplesPer10Ms));
117
118  FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
119                              sample_rate);
120
121  rtc::scoped_ptr<int8_t[]> out_buffer;
122  out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
123  rtc::scoped_ptr<int8_t[]> in_buffer;
124  in_buffer.reset(new int8_t[kFrameSizeBytes]);
125  for (int i = 0; i < kNumberOfFrames; ++i) {
126    fine_buffer.GetPlayoutData(out_buffer.get());
127    EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
128    UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
129    fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
130  }
131}
132
133TEST(FineBufferTest, BufferLessThan10ms) {
134  const int kSampleRate = 44100;
135  const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
136  const int kFrameSizeSamples = kSamplesPer10Ms - 50;
137  RunFineBufferTest(kSampleRate, kFrameSizeSamples);
138}
139
140TEST(FineBufferTest, GreaterThan10ms) {
141  const int kSampleRate = 44100;
142  const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
143  const int kFrameSizeSamples = kSamplesPer10Ms + 50;
144  RunFineBufferTest(kSampleRate, kFrameSizeSamples);
145}
146
147}  // namespace webrtc
148